
/**
 * The MIT License (MIT)
 *
 * Copyright (c) 2013-2021 Winlin
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy of
 * this software and associated documentation files (the "Software"), to deal in
 * the Software without restriction, including without limitation the rights to
 * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
 * the Software, and to permit persons to whom the Software is furnished to do so,
 * subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in all
 * copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
 * FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
 * COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
 * IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
 * CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 */
 import axios from 'axios'
export default {
// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
// 基于SRS RTC发布服务器的异步awat prmise。
    SrsRtcPublisherAsync() {
        var self = {};
        self.constraints = {
            audio: true,
            video: {
                width: {ideal: 320, max: 576}
            }
        };
        self.publish = async function (url) {
            var conf = self.__internal.prepareUrl(url);
            self.pc.addTransceiver("audio", {direction: "sendonly"});
            self.pc.addTransceiver("video", {direction: "sendonly"});
            var stream = await navigator.mediaDevices.getUserMedia(self.constraints);
            // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
            stream.getTracks().forEach(function (track) {
                self.pc.addTrack(track);

                // Notify about local track when stream is ok.
                self.ontrack && self.ontrack({track: track});
            });

            var offer = await self.pc.createOffer();
            await self.pc.setLocalDescription(offer);
            var session = await new Promise(function (resolve, reject) {
                // @see https://github.com/rtcdn/rtcdn-draft
                var data = {
                    api: conf.apiUrl, tid: conf.tid, streamurl: conf.streamUrl,
                    clientip: null, sdp: offer.sdp
                };
                axios.post(conf.apiUrl, JSON.stringify(data),{
                    headers:{
                        'content-type':'application/json'
                    }
                }).then((res) => {
                    if (res.data.code) {
                        reject(res.data); return;
                    }
                    resolve(res.data);
                }).catch((err) => {
                    reject(err);
                })
            });
            await self.pc.setRemoteDescription(
                new RTCSessionDescription({type: 'answer', sdp: session.sdp})
            );
            session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';

            return session;
        };

        // Close the publisher.
        self.close = function () {
            self.pc && self.pc.close();
            self.pc = null;
        };

        // The callback when got local stream.
        // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
        self.ontrack = function (event) {
            // Add track to stream of SDK.
            self.stream.addTrack(event.track);
        };

        // Internal APIs.
        self.__internal = {
            defaultPath: '/rtc/v1/publish/',
            prepareUrl: function (webrtcUrl) {
                var urlObject = self.__internal.parse(webrtcUrl);

                // If user specifies the schema, use it as API schema.
                var schema = urlObject.user_query.schema;
                schema = schema ? schema + ':' : window.location.protocol;

                var port = urlObject.port || 1985;
                if (schema === 'https:') {
                    port = urlObject.port || 443;
                }

                // @see https://github.com/rtcdn/rtcdn-draft
                var api = urlObject.user_query.play || self.__internal.defaultPath;
                if (api.lastIndexOf('/') !== api.length - 1) {
                    api += '/';
                }

                apiUrl = schema + '//' + urlObject.server + ':' + port + api;
                for (var key in urlObject.user_query) {
                    if (key !== 'api' && key !== 'play') {
                        apiUrl += '&' + key + '=' + urlObject.user_query[key];
                    }
                }
                // Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
                var apiUrl = apiUrl.replace(api + '&', api + '?');

                var streamUrl = urlObject.url;

                return {
                    apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port,
                    tid: Number(parseInt(new Date().getTime()*Math.random()*100)).toString(16).substr(0, 7)
                };
            },
            parse: function (url) {
                // @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
                var a = document.createElement("a");
                a.href = url.replace("rtmp://", "http://")
                    .replace("webrtc://", "http://")
                    .replace("rtc://", "http://");

                var vhost = a.hostname;
                var app = a.pathname.substr(1, a.pathname.lastIndexOf("/") - 1);
                var stream = a.pathname.substr(a.pathname.lastIndexOf("/") + 1);

                // parse the vhost in the params of app, that srs supports.
                app = app.replace("...vhost...", "?vhost=");
                if (app.indexOf("?") >= 0) {
                    var params = app.substr(app.indexOf("?"));
                    app = app.substr(0, app.indexOf("?"));

                    if (params.indexOf("vhost=") > 0) {
                        vhost = params.substr(params.indexOf("vhost=") + "vhost=".length);
                        if (vhost.indexOf("&") > 0) {
                            vhost = vhost.substr(0, vhost.indexOf("&"));
                        }
                    }
                }

                // when vhost equals to server, and server is ip,
                // the vhost is __defaultVhost__
                if (a.hostname === vhost) {
                    var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
                    if (re.test(a.hostname)) {
                        vhost = "__defaultVhost__";
                    }
                }

                // parse the schema
                var schema = "rtmp";
                if (url.indexOf("://") > 0) {
                    schema = url.substr(0, url.indexOf("://"));
                }

                var port = a.port;
                if (!port) {
                    if (schema === 'http') {
                        port = 80;
                    } else if (schema === 'https') {
                        port = 443;
                    } else if (schema === 'rtmp') {
                        port = 1935;
                    }
                }

                var ret = {
                    url: url,
                    schema: schema,
                    server: a.hostname, port: port,
                    vhost: vhost, app: app, stream: stream
                };
                self.__internal.fill_query(a.search, ret);

                // For webrtc API, we use 443 if page is https, or schema specified it.
                if (!ret.port) {
                    if (schema === 'webrtc' || schema === 'rtc') {
                        if (ret.user_query.schema === 'https') {
                            ret.port = 443;
                        } else if (window.location.href.indexOf('https://') === 0) {
                            ret.port = 443;
                        } else {
                            // For WebRTC, SRS use 1985 as default API port.
                            ret.port = 1985;
                        }
                    }
                }

                return ret;
            },
            fill_query: function (query_string, obj) {
                // pure user query object.
                obj.user_query = {};

                if (query_string.length === 0) {
                    return;
                }

                // split again for angularjs.
                if (query_string.indexOf("?") >= 0) {
                    query_string = query_string.split("?")[1];
                }

                var queries = query_string.split("&");
                for (var i = 0; i < queries.length; i++) {
                    var elem = queries[i];

                    var query = elem.split("=");
                    obj[query[0]] = query[1];
                    obj.user_query[query[0]] = query[1];
                }

                // alias domain for vhost.
                if (obj.domain) {
                    obj.vhost = obj.domain;
                }
            }
        };

        self.pc = new RTCPeerConnection(null);

        // To keep api consistent between player and publisher.
        // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
        // @see https://webrtc.org/getting-started/media-devices
        self.stream = new MediaStream();

        return self;
    },

// Depends on adapter-7.4.0.min.js from https://github.com/webrtc/adapter
// 基于异步等待承诺的SRS RTC播放器
    SrsRtcPlayerAsync() {
        var self = {};
        // @see https://github.com/rtcdn/rtcdn-draft
        // @url The WebRTC url to play with, for example:
        //      webrtc://r.ossrs.net/live/livestream
        // or specifies the API port:
        //      webrtc://r.ossrs.net:11985/live/livestream
        // or autostart the play:
        //      webrtc://r.ossrs.net/live/livestream?autostart=true
        // or change the app from live to myapp:
        //      webrtc://r.ossrs.net:11985/myapp/livestream
        // or change the stream from livestream to mystream:
        //      webrtc://r.ossrs.net:11985/live/mystream
        // or set the api server to myapi.domain.com:
        //      webrtc://myapi.domain.com/live/livestream
        // or set the candidate(ip) of answer:
        //      webrtc://r.ossrs.net/live/livestream?eip=39.107.238.185
        // or force to access https API:
        //      webrtc://r.ossrs.net/live/livestream?schema=https
        // or use plaintext, without SRTP:
        //      webrtc://r.ossrs.net/live/livestream?encrypt=false
        // or any other information, will pass-by in the query:
        //      webrtc://r.ossrs.net/live/livestream?vhost=xxx
        //      webrtc://r.ossrs.net/live/livestream?token=xxx
        self.play = async function(url) {
            var conf = self.__internal.prepareUrl(url);
            self.pc.addTransceiver("audio", {direction: "recvonly"});
            self.pc.addTransceiver("video", {direction: "recvonly"});

            var offer = await self.pc.createOffer();
            await self.pc.setLocalDescription(offer);
            var session = await new Promise(function(resolve, reject) {
                // @see https://github.com/rtcdn/rtcdn-draft
                var data = {
                    api: conf.apiUrl, tid: conf.tid, streamurl: conf.streamUrl,
                    clientip: null, sdp: offer.sdp
                };
                
                axios.post(conf.apiUrl, JSON.stringify(data),{
                    headers:{
                        'content-type':'application/json'
                    },
                }).then((res) => {
                    if (res.data.code) {
                        reject(res.data); return;
                    }
                    resolve(res.data);
                }).catch((err) => {
                    reject(err);
                })
            });
            await self.pc.setRemoteDescription(
                new RTCSessionDescription({type: 'answer', sdp: session.sdp})
            );
            session.simulator = conf.schema + '//' + conf.urlObject.server + ':' + conf.port + '/rtc/v1/nack/';
            return session;
        };

        // Close the player.
        self.close = function() {
            self.pc && self.pc.close();
            self.pc = null;
        };

        // The callback when got remote track.
        // Note that the onaddstream is deprecated, @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/onaddstream
        self.ontrack = function (event) {
            // https://webrtc.org/getting-started/remote-streams
            self.stream.addTrack(event.track);
        };

        // Internal APIs.
        self.__internal = {
            defaultPath: '/rtc/v1/play/',
            prepareUrl: function (webrtcUrl) {
                var urlObject = self.__internal.parse(webrtcUrl);
                // If user specifies the schema, use it as API schema.
                var schema = urlObject.user_query.schema;
                schema = schema ? schema + ':' : window.location.protocol;

                var port = urlObject.port || 1985;
                if (schema === 'https:') {
                    port = urlObject.port || 443;
                }
                // @see https://github.com/rtcdn/rtcdn-draft
                var api = urlObject.user_query.play || self.__internal.defaultPath;
                if (api.lastIndexOf('/') !== api.length - 1) {
                    api += '/';
                }

                apiUrl = schema + '//' + urlObject.server + ':' + port + api;
                for (var key in urlObject.user_query) {
                    if (key !== 'api' && key !== 'play') {
                        apiUrl += '&' + key + '=' + urlObject.user_query[key];
                    }
                }
                // Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
                var apiUrl = apiUrl.replace(api + '&', api + '?');

                var streamUrl = urlObject.url;

                return {
                    apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port,
                    tid: Number(parseInt(new Date().getTime()*Math.random()*100)).toString(16).substr(0, 7)
                };
            },
            parse: function (url) {
                // @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
                var a = document.createElement("a");
                a.href = url.replace("rtmp://", "http://")
                    .replace("webrtc://", "http://")
                    .replace("rtc://", "http://");

                var vhost = a.hostname;
                var app = a.pathname.substr(1, a.pathname.lastIndexOf("/") - 1);
                var stream = a.pathname.substr(a.pathname.lastIndexOf("/") + 1);

                // parse the vhost in the params of app, that srs supports.
                app = app.replace("...vhost...", "?vhost=");
                if (app.indexOf("?") >= 0) {
                    var params = app.substr(app.indexOf("?"));
                    app = app.substr(0, app.indexOf("?"));

                    if (params.indexOf("vhost=") > 0) {
                        vhost = params.substr(params.indexOf("vhost=") + "vhost=".length);
                        if (vhost.indexOf("&") > 0) {
                            vhost = vhost.substr(0, vhost.indexOf("&"));
                        }
                    }
                }

                // when vhost equals to server, and server is ip,
                // the vhost is __defaultVhost__
                if (a.hostname === vhost) {
                    var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
                    if (re.test(a.hostname)) {
                        vhost = "__defaultVhost__";
                    }
                }

                // parse the schema
                var schema = "rtmp";
                if (url.indexOf("://") > 0) {
                    schema = url.substr(0, url.indexOf("://"));
                }

                var port = a.port;
                if (!port) {
                    if (schema === 'http') {
                        port = 80;
                    } else if (schema === 'https') {
                        port = 443;
                    } else if (schema === 'rtmp') {
                        port = 1935;
                    }
                }

                var ret = {
                    url: url,
                    schema: schema,
                    server: a.hostname, port: port,
                    vhost: vhost, app: app, stream: stream
                };
                self.__internal.fill_query(a.search, ret);

                // For webrtc API, we use 443 if page is https, or schema specified it.
                if (!ret.port) {
                    if (schema === 'webrtc' || schema === 'rtc') {
                        if (ret.user_query.schema === 'https') {
                            ret.port = 443;
                        } else if (window.location.href.indexOf('https://') === 0) {
                            ret.port = 443;
                        } else {
                            // For WebRTC, SRS use 1985 as default API port.
                            ret.port = 1985;
                        }
                    }
                }

                return ret;
            },
            fill_query: function (query_string, obj) {
                // pure user query object.
                obj.user_query = {};

                if (query_string.length === 0) {
                    return;
                }

                // split again for angularjs.
                if (query_string.indexOf("?") >= 0) {
                    query_string = query_string.split("?")[1];
                }

                var queries = query_string.split("&");
                for (var i = 0; i < queries.length; i++) {
                    var elem = queries[i];

                    var query = elem.split("=");
                    obj[query[0]] = query[1];
                    obj.user_query[query[0]] = query[1];
                }

                // alias domain for vhost.
                if (obj.domain) {
                    obj.vhost = obj.domain;
                }
            }
        };
        self.pc = new RTCPeerConnection(null);
        // Create a stream to add track to the stream, @see https://webrtc.org/getting-started/remote-streams
        self.stream = new MediaStream();
        // https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/ontrack
        self.pc.ontrack = function(event) {
            if (self.ontrack) {
                self.ontrack(event);
            }
        };

        return self;
    },

    // Format the codec of RTCRtpSender, kind(audio/video) is optional filter.
    // https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
    SrsRtcFormatSenders(senders, kind) {
        var codecs = [];
        senders.forEach(function (sender) {
            var params = sender.getParameters();
            params && params.codecs && params.codecs.forEach(function(c) {
                if (kind && sender.track.kind !== kind) {
                    return;
                }

                if (c.mimeType.indexOf('/red') > 0 || c.mimeType.indexOf('/rtx') > 0 || c.mimeType.indexOf('/fec') > 0) {
                    return;
                }

                var s = '';

                s += c.mimeType.replace('audio/', '').replace('video/', '');
                s += ', ' + c.clockRate + 'HZ';
                if (sender.track.kind === "audio") {
                    s += ', channels: ' + c.channels;
                }
                s += ', pt: ' + c.payloadType;

                codecs.push(s);
            });
        });
        return codecs.join(", ");
    }
}